Looks very good for active speakers with the built in cross-over.
Are any products ready using this?

/Eric

I can't name names because of confidentiality but I know Zetex have got DDFA customers signed up for mass production this year.

From what I heard of their prototypes the sound is excellent, and everyone who's heard them seems to agree with this.

I do know that the measured performance is pretty astounding with noise and distortion more than 110dB down from full scale -- remember this is digital in not just a power amp, so it includes the functionality of a DAC (and DSP) as well.

One of their target markets is indeed active speakers, the DDFA chip includes everything needed to do all this digitally (crossover, equalisation, limiting, fault protection...).

The big issue with DDFA for high-power amplifiers (like PA which I'm interested in but is not their main target area) is the high switching speed (844kHz -- needed for digital noise shaping) which makes it very difficult to get high efficiency.

However I've been looking into this in some detail and reckon that by using fast-switching devices (+/-190V half-bridge using CoolMOS CS + series Schottky + SiC reverse diode) the loss can be kept down to a few percent even at this speed -- though it's certainly not at all easy switching 35A through 380V in less than 10ns it can be done, but extreme care is needed to avoid di/dt and dv/dt problems (nice radio transmitter...)

The big issue with DDFA for high-power amplifiers (like PA which I'm interested in but is not their main target area) is the high switching speed (844kHz -- needed for digital noise shaping) which makes it very difficult to get high efficiency.

You can find it at http://www.diyaudio.com/forums/showt...52#post1583852 .
The problem comes from the infinite spectra of the PWM is being sampled by the digital amplifier on its output. It can cause spectral overlap, which appears as distortion because the distances of the non-zero energy frequencies beside the carrier harmonics are integral multiples of the modulating signal's frequecy.

Originally posted by Gyula You can find it at http://www.diyaudio.com/forums/showt...52#post1583852 .
The problem comes from the infinite spectra of the PWM is being sampled by the digital amplifier on its output. It can cause spectral overlap, which appears as distortion because the distances of the non-zero energy frequencies beside the carrier harmonics are integral multiples of the modulating signal's frequecy.

You don't understand how all-digital PWM works in circuits like this. The digital sample-value-to-pulsewidth conversion (with about 9ns time resolution) produces quantisation error which is then noise-shaped to push it out of the audio band, but there's no intermodulation or aliasing of PWM harmonics because the input signal is already sampled and all processing is discrete-time -- there are no "infinite bandwidth" signals.

The difference between the amplifier output and an "ideal" reference DAC (<-120dB THD+N) is integrated and fed back into the PWM modulator -- this is equivalent to wideband negative feedback in an analogue amplifier.

The total noise+distortion at the DDFA amp output -- including all harmonics and spurious components -- is better than most standalone DACs, so the power amplifier function adds little or no signal degradation compared to a simple DAC.

You don't understand how all-digital PWM works in circuits like this. The digital sample-value-to-pulsewidth conversion (with about 9ns time resolution) produces quantisation error which is then noise-shaped to push it out of the audio band, but there's no intermodulation or aliasing of PWM harmonics because the input signal is already sampled and all processing is discrete-time -- there are no "infinite bandwidth" signals.

I suggest you to learn the very basics of describing the signals in frequency domain. Then consider that even a discrete-time signal has infinite spectra. Btw. what do you think about for example why a ZOH is needed at the output of a simple DAC?

After that I think you should make some calculation of the spectra of a non-D.C. modulated PWM signal. For example what do you think why is a Low-pass filter needed at the output of Class-D?

Please tell me how can you expect linear operation from a non-linear process?

Have you ever heard about that sampling has to be done on bandwith-limited input signal?

Have you ever watched the sway on a filtered PWM output? Do you think if you sample that then it has to be equal to the input samples only because it's sampled?

Is the sampling a wonder or some kind of magic in your opinion? Or what?

I suggest you to learn the very basics of describing the signals in frequency domain. Then consider that even a discrete-time signal has infinite spectra. Btw. what do you think about for example why a ZOH is needed at the output of a simple DAC?

After that I think you should make some calculation of the spectra of a non-D.C. modulated PWM signal. For example what do you think why is a Low-pass filter needed at the output of Class-D?

Please tell me how can you expect linear operation from a non-linear process?

Have you ever heard about that sampling has to be done on bandwith-limited input signal?

Have you ever watched the sway on a filtered PWM output? Do you think if you sample that then it has to be equal to the input samples only because it's sampled?

Is the sampling a wonder or some kind of magic in your opinion? Or what?

I know exactly how PWM works, but this is a sampled-data digital system *not* an analogue PWM system which can suffer from intermodulation sidebands.

The input is sampled digital data, this is upconverted and then converted to a discrete-time quantised PWM signal where the location of all the sample edges falls on a 108MHz grid. Any quantisation error due to this time resolution is then noise-shaped so it appears above the audio band.

If you look at the spectrum of the result, it's almost identical to a noise-shaped non-PWM sigma-delta DAC -- which is where the analogue signal input to a conventional power amplifier would most likely come from. The output needs lowpass filtering in exactly the same way that a conventional DAC does and for the same reasons. There are no nonlinear effects from the fast risetime edges any more than there are in the stepped output from a conventional DAC.

Before suggesting I learn more about such things you might like to consider that I've spent the last twenty years or so designing complex mixed-signal ICs, including some of the highest performance ADCs and DACs on the planet -- and I'm talking *much* higher performance than most audio here, with jitter down to a fraction of a picosecond and sampling rates up to tens of GHz...